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    8,362 send voice flash media server sip робіт знайдено, ціни вказані в USD
    Callcenter Application Закінчився left

    ...-Search CDR - with the phone number i can see a record when its called and how many time its called by the dialer. -Download Recordings - I can download here records from the server. -Recordings - i can listen here the whole conversation and even edit the resultcode of this client (what i mean with edit is example: From sale i can set it to Negative or

    $3080 (Avg Bid)
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    Twilio SIP set-up Закінчився left

    Please set-up Twilio phone number with SIP so I can use on following site: [увійдіть, щоб побачити URL]

    $19 (Avg Bid)
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    VOIP - CALL DROPS Закінчився left

    We use a XORCOM ELASTIX VOIP and have 2 office locations with a VPN tunnel. The second office cant make or receive calls which seems to be a SIP ALG or NAT issue. But need this fixed.

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    OBD CALLING Закінчився left

    Hello We have SIP,PRI and want to build script where we can make OBD Calling (asterisk),

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    I have a working FreePBX server. The freepbx server is running good so far. The following changes needed in my server. 1. I want to install a open source predictive Dialer in my freepbx. I have have chosen VICIDIAL for that. You may suggest better one. The dialer must use existing extensions for auto dialing features. 2. You must configure the system

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    ...provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [увійдіть, щоб побачити URL]: Caller I...

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    Project for Bhavin B. Закінчився left

    ...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed

    $37 / hr (Avg Bid)
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    1 заявки

    We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization

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    App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.

    $642 (Avg Bid)
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    Provisioning cloud Cisco 7911 Закінчився left

    Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[увійдіть, щоб побачити URL]" para provisionamiento remoto.

    $130 (Avg Bid)
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    4 заявки

    Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.

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    ...Prevost, Quebec) Tax Rule rate = 14.98% 2min outbound call on SIP Canuk 200 plan 0.020 = 6sec increment 120sec 2min * 0 .020 * 1.1498 = 0.045992 (round up to 0.045) 3min inbound call on SIP Canuk 200 plan 0.025 = 6sec increment 180sec 3min * 0 .025 * 1.1498 = 0.086235 (round up to 0.086) SIP Canuk 200 Package Detail [увійдіть, щоб побачити URL] & [увійдіть, щоб побачи...

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    Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own extensions and modify their call routing. I need a developer that is an expert in Node.js as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with

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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well

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    VLC and FreePBX on Ubuntu Закінчився left

    VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist

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    Looking For Android sip app Devloper Закінчився left

    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [увійдіть, щоб побачити URL] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [увійдіть, щоб побачити URL] will be long term project if satisfied with the [увійдіть, щоб побачити URL] budget i...

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    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [увійдіть, щоб побачити URL] /hangupall [увійдіть, щоб побачити URL] /answer [увійдіть, щоб побачити URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [увійдіть, щоб побачити URL] msip:hangupall [увійдіть, щоб побачити URL] msip:answer [увійдіть, щоб п...

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    want build code using nmap Закінчився left

    hello, please check this doc [увійдіть, щоб побачити URL] i need code to test my back Udp + Tcp connection to my sip server with show error Error = { TRYING = 100, RING = 180, TIMEOUT = 408, BUSY = 486, DECLINE = 603, OK = 200, UNAUTHORIZED = 401, FORBIDDEN = 403, NOTFOUND = 404, PROXY_AUTH_REQUIRED =

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    VOIP/SIP Project Закінчився left

    ~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at...(Sydney, Hong Kong - new at WeWork, China) Looking at setup cloud solution to connect existing cisco/ gransdstream sip phones. Currently using Faktortel from Australia - looking at Freeswitch/ asterisk + Twilio SIP trunk (HK phase 1) + Faktortel sip trunk (AU phase 2).

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    Skype for business Direct Routing Закінчився left

    We are looking for a skype for business office365 consultant to assist in setting up a direct routing environment from the office365 environment to a SIP SBC.

    $39 / hr (Avg Bid)
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    ...and we can organise it with AWS) to: a. Connects to a Client SIP Trunk (We will use a Hosted PABX to test) b. Connects to a Carrier SIP Trunk (We will use a Hosted PABX to test) c. Passes calls between the Client and the Carrier d. Will hide the IP of both Media and Signalling e. Server will be built on AWS 2. Assist and provide basic training /

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    I am looki...connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration document (outlining works completed) Initial configuration could lead to additional future works.

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    I'm searching for a multithreaded perl script that calls a phonennumber mutiple times with predefined sip-accounts and 4 additional functions.

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    Sip dialer voice recorder coding Закінчився left

    Sip dialer voice recorder coding Must know about PHP and .net programming. Experienced person.

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    Help setting up Freepbx Закінчився left

    i have set up a new freepbx server it is online but i cannot get the SIP registration to go through and actually connect to the provider this is not a big project i am missing a setting somewhere and i need someone to connect to my server via team viewer and fix it for me

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    SIP to Local call App Закінчився left

    Need to create a SIP app that can install on Android. Register to SIP server Receive call from SIP server and initiate call via telephone sim card Connect audio from telephone call to SIP call

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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

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    I have a 3CX server hosted on Azure with a public IP address. However, the client IP phones are behind NAT and common NAT traversal techniques such as STUN and TURN are cannot be used. I am configuring an outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions:

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    install FreePBX v.13.0.121+; Закінчився left

    1. Install FreePBX v.13.0.121+ in to Linode server 2. configure SIP trunk engin (Australia) , (can be copy from existing PABX) 3. Connect PSTN to sip using ATA adaptor 4 Connect with Britex 24

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    Looking to have a application that registers to a sip server based on sip 2.0 and or direct ip calling have the ability to make calls. open source can be used. up to 10 favorite contacts can be created in tile format. does not need to incorportate with native android phone contacts(for now) each contact(favorite should have a place to put in rtsp string

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    SIP Media Gateway Engine for Chatbot Закінчився left

    ...bring voice to text-based chatbots. The Gateway accepts SIP VoIP and then communicates with a chatbot over a JSON/REST interface. The Gateway also communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711 and G.722 RTP media interface

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    Обраний Терміновий Угода про нерозголошення
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    linphone + uvc camera Закінчився left

    I would like to use the uvc camera on my linphone app to make video connections. If the ...would like to use the uvc camera on my linphone app to make video connections. If the linphone app gives you the source that is normally built, register it according to the sip information I provided and make the video connection possible through the uvc camera.

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    ...using Technology such as NODE.js/ React native app and react native web When building the app like whatsapp we need to make sure that all in network calls stay on the app server , we a place on the admin to add all the millions of numbers we have in our network now and the ability to keep adding to the list very easily. Any calls that are going

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    Asterisk Black listing Закінчився left

    ...calls with the same numbers (robot calls). We are starting a wholesale VoIP business, but we have some clients that if we have 8 or let say 24 ports open, some times they will send 8 or 24 calls depending on the ports available and all ports will be ringing at the same time with the same number let say (212-347-2222). we want to find a solution to the

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    Android with SIP/VoIP caller program Закінчився left

    ...develop an Android Studio application that allows to include the SIP / VOIP SDK to connect to ASTERISK The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module More details (ask) ////////

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    ...APPBuilder that allows to include the SIP / VOIP SDK to connect to ASTERISK as an option and if it were not possible in ANDROID Studio The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module

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    Trophy icon Content Writing Закінчився left

    ... Its all about Fun Art, not Fine Art! And remember, we are a BYO studio, so bring along your favourite bottle or two of liquid creativity to enjoy while you paint and sip with us on the day. You supply the wine and we’ll provide all the art materials, glassware and good vibes. Leave the rest to us! This is not your typical painting class

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    Paint and sip for kids Закінчився left

    looking for an artist/painter/instructor to teach kids and art classes/parties for different ages. 3-5 hours a day, 2 times a week. Ideal candidate should be positive, patient, energetic. Looking to hire immediately.

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    Місцевий
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    Vicidial webform Закінчився left

    ...gin=gs102&phone_pass=865748&fronter=321&closer=321&group=MORTCP&channel_group=MORTCP&SQLdate=2018-05-22+113857&epoch=1527014338&uniqueid=1527014305.751&customer_zap_channel=SIP/enterprise-000000ff&customer_server_ip=&server_ip=69.64.71.182&SIPexten=gs102&session_id=8600056&phone=7029973001&parked_by=55502&dispo=&dialed_n...

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    Asterisk Management System with PHP Закінчився left

    ...not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers - Create

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    Asterisk dialplan Закінчився left

    ...can be achieved with a call conference or call bridging. when dialing user (A) calls user (B) they are put into a call conference or bridge. if the dialer(user a) hangs up(sip 487) before user b answers, a disconnects but the conference / bridge channel stays open until b hangs up or for a max of 30 seconds cli needs to be passed from a to be etc

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    Moile Sip App coding Закінчився left

    We need sip protocol writer. There is sip application for testing for voice also need recording of voice on app

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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio in chunks to google speech 3. get the recognized text back I would like to either detect the silence to send the audio to speech service or continuous send and get the text as it is ready

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    Install IP PBX Закінчився left

    ...mutually). 2. Configure TLS/SRTP to traverse voice over ssl. 3. Ensure that traffic is passed through blocked gateways. 4. Configure auto provisioning for IP Phones and mobile clients. 5. Mobile clients must be compatible across various OS. 6. Configure SIP trunk for International calls (preferably google voice). 7. Any other work that shall be required

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    Trophy icon Company Website - System Integration with XML-RPC API Закінчився left

    We are a VoIP company selling SIP trunks and Hosted PBX services. We are looking for a new modern website for our company to showcase our products and services. The clients should be able to do some basic account management on our website where the API side comes to our SIP server. Only the basics like viewing account balance and topup account with

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    We need a Phone Sales Person (English Speaking Countries) Part time for at least 20h/week. PEQRFECT ENGLISH IS A REQUIREMENT. You will work with our CRM / SIP Phone.

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    Trophy icon Design a fun poster for our Paint Studio Закінчився left

    We would like a poster designed which will be hung on the wall of our Paint and Sip studio. The poster should be 300 dpi/high resolution as it will be printed out at about 45 inch wide to 90 inch high (so thats the ratio on size ) and hung on our wall. The poster shows a list of fun things that people might do at our studio while at our events. So

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    i want create my own website Закінчився left

    i want to create own website like mutual fund , sip and financial and admin login client login and sip calculator also available in my web site and nse,mcx stock market api also required

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    I'm looking for someone to develop an Outlook plugin for windows that you login with a VoIP SIP extension credentials and whenever a call comes in it searches the number in Outlook and if it finds a contact it opens up that contact in a new window. If it doesn't find a contact it opens a new contact window with filed in the details from the caller ID

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